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How To JIO SIP trunk asterisk vicidial freepbx

Estimated read time: 2 min

Topic : Step by step document to configure JIO sip trunk in asterisk (vicidial/freepbx)

Description: Reliance JIO SIP Trunk configuration in asterisk based Telephony servers.


About JIO SIP trunk :

           Reliance Jio Leading telecom vendor in INDIA, who also provides ISDN/E1 trunk via Ethernet in SIP Protocol (ie VOIP) to customer similar to SIP trunk provided by TATA/AIRTEL.
As shown in the picture below JIO provides a dedicate network to connect to there SIP gateway to register the SIP trunk.
Either you need to have dedicate ethernet port in asterisk server to connect to jio network, or redesign the network setup to opt to JIO subnet.

Reliance jio sip trunk asterisk


Details provided by Jio:

            Once you have purchased  the new jio sip trunk, you will be provided with below details
1. DID numbers
2. Pilot number
3. SIP gateway IP
4. Jio Network subnet range .


Step 1: Network Connection

            As show in picture above ,connect your Asterisk server to the JIO network and assign the IP address from the subnet provided by jio either to a second ethernet or existing ethernet port(based on your network design).
The network configuration is not covered here , it might defer between OS type you are using, but concept is same.

Step 2:  Network route to JIO gateway (optional)
       If you are connecting JIO to your existing networks as pre the picture shown, then you might need a dedicate route to reach the JIO gateway and also media ip.
Strict RTP switching to RTP target address 
Check your OS network settings to set static route to jio network.

Step 3: SIP trunk peer settings.

        Once you have setup the network , make sure you are able to reach the JIO gateway and SIP gateway ip by ping command.
       enter the below sip settings in you asterisk sip.conf,  if are using Freepbx you can create the same in Gui trunk settings, similar for vicidial/goautodial under carrier settings.


[jiosiptrunk]
type=friend
disallow=all
allow=alaw
allow=ulaw
allow=g729
host=100.XXX.XXX.XXX ;this is jio sip gateway ip
fromdomain=100.XXX.XXX.XXX 
;this is jio sip gateway ip
qualify=yes
dtmfmode=rfc2833
nat=no
context=from-trunk  ;context to receive inbound calls
sendrpid=yes
trustrpid=yes


Step 4: Dialplan to dial out

Note: The key factor to dial out via JIO , you need to set proper callerid while dialing out.

Use the below dialplan , alter according to your requirement.
For vanilla Asterisk :

exten => _X.,1,Set(CALLERID(num)=+914412345678)
;4412345678 is my pilot no, you have to enter your pilot no or any number from the DID range
exten => _X.,2,Dial(SIP/jiosiptrunk/${EXTEN})
exten => _X.,3,Hangup

Vicidial / Goautodial

exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Set(CALLERID(num)=4412345678)
;4412345678 is my pilot no, you have to enter your pilot no or any number from the DID range
exten => _X.,3,Dial(SIP/jiosiptrunk/${EXTEN},,tTo)
exten => _X.,4,Hangup

Freepbx 

For Freepbx, you need create outbound route , with necessary Dial pattern and make sure to set the ROUTE CID (either pilot or DID numbers)
if  you want individual CID for each user make sure each user extension is configured with outbound callerid with there DID number. 


Conclusion:
Hope the above tutorial is helpful to configure the JIO siptrunk in your sip server.
For professional support reach me via skype: striker24x7  mail; striker24x7@gmail.com

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